U.S. patent application number 13/504680 was filed with the patent office on 2012-08-30 for speech enhancement method and system.
This patent application is currently assigned to Phonak AG. Invention is credited to Samuel Harsch.
Application Number | 20120221329 13/504680 |
Document ID | / |
Family ID | 41466376 |
Filed Date | 2012-08-30 |
United States Patent
Application |
20120221329 |
Kind Code |
A1 |
Harsch; Samuel |
August 30, 2012 |
SPEECH ENHANCEMENT METHOD AND SYSTEM
Abstract
A method of speech enhancement in a room (10) includes the steps
of capturing audio signals from a speaker's voice by a microphone
(12), estimating an ambient noise level in the room from the
captured audio signals, processing the captured audio signals by an
audio signal processing unit (20), estimating a reverberation
level, determining the gain to be applied to the captured audio
signals by the audio signal processing unit according to a
comparison between the estimated ambient noise level and the
estimated reverberation level, and generating sound according to
the processed audio signals by a loudspeaker arrangement (24)
located in the room, wherein the reverberation level is the level
of reverberant components of the sound generated by the loudspeaker
arrangement.
Inventors: |
Harsch; Samuel; (Ballaigues,
CH) |
Assignee: |
Phonak AG
Staefa
CH
|
Family ID: |
41466376 |
Appl. No.: |
13/504680 |
Filed: |
October 27, 2009 |
PCT Filed: |
October 27, 2009 |
PCT NO: |
PCT/EP2009/064142 |
371 Date: |
April 30, 2012 |
Current U.S.
Class: |
704/225 ;
704/E19.039; 704/E21.002 |
Current CPC
Class: |
H04R 3/02 20130101; H04S
7/305 20130101; H04S 7/301 20130101 |
Class at
Publication: |
704/225 ;
704/E19.039; 704/E21.002 |
International
Class: |
G10L 21/02 20060101
G10L021/02; H04B 3/20 20060101 H04B003/20; G10L 19/14 20060101
G10L019/14 |
Claims
1-25. (canceled)
26. A method of speech enhancement in a room, comprising capturing
audio signals from a speaker's voice by a microphone, estimating an
ambient noise level in the room from the captured audio signals,
processing the captured audio signals by an audio signal processing
unit, estimating a reverberation level, determining a gain to be
applied to the captured audio signals by the audio signal
processing unit according to a comparison between the estimated
ambient noise level and the estimated reverberation level, and
generating sound according to the processed audio signals by a
loudspeaker arrangement located in the room, wherein the
reverberation level is the level of reverberant components of the
sound generated by the loudspeaker arrangement.
27. The method of claim 26, wherein the reverberation level is
estimated from a level of the processed audio signals or from a
level of the audio signals supplied to audio signal processing
unit.
28. The method of claim 27, wherein the processed audio signal
undergo amplification at constant gain by a power amplifier prior
to being supplied as input to the loudspeaker arrangement as
amplified processed audio signals.
29. The method of claim 26, comprising the further step of
determining whether the speaker is presently speaking or not from
the captured audio signals using a voice activity detector, and
wherein the ambient noise level is estimated from a level of the
audio signals captured during times when it has been determined
that the speaker is not speaking.
30. The method of claim 29, wherein, during times when it has been
determined that the speaker is speaking, the gain is increased to a
level at which the ambient noise level is expected to be masked by
the reverberation level.
31. The method of claim 30, wherein the gain is limited to a
maximum value corresponding to a gain at which the reverberation
level exceeds the ambient noise level by a given threshold
value.
32. The method of claim 31, wherein the threshold value is 3
dB.
33. The method of claim 26, wherein it is determined, by a feedback
canceller, whether a gain applied by the audio signal processing
unit causes a critical feedback level, and wherein, when a critical
feedback level has been determined, the gain applied by the audio
signal processing unit is limited to values which do not cause a
critical feedback level.
34. The method of claim 26, wherein the reverberation level is
estimated from a level of the processed audio signals by using
acoustic room parameters.
35. The method of claim 34, wherein the reverberation level is
estimated from a level of the processed audio signals by applying a
correction factor derived from the acoustic room parameters to a
level measurement at an input of the power amplifier.
36. The method of claim 34, wherein the acoustic room parameters
are fixed and are that of a room having characteristics similar to
those expected to exist in the room in which the loudspeaker
arrangement is to be used.
37. The method of claim 34, wherein the acoustic room parameters
are determined in-situ in a calibration mode prior to starting
speech enhancement operation.
38. The method of claim 37, wherein the acoustic room parameters
are determined by measurement of a level of the reverberant field
in the room.
39. The method of claim 38, wherein, in the calibration mode, the
microphone is placed at a position in the room which is dominated
by reverberant sound from the loudspeaker arrangement, a test
signal with a known level is generated via the loudspeaker
arrangement, the test signal is captured by the microphone, and a
correction factor is determined from a level of the test audio
signals captured by the microphone.
40. The method of claim 37, wherein the acoustic room parameters
are determined by measurement of an impulse response of the
room.
41. The method of claim 40, wherein, in the calibration mode, the
microphone is placed at any position in the room, a maximum length
sequence test signal is generated at a known level via the
loudspeaker arrangement, the test signal is captured by the
microphone, and a correction factor is determined from a level of
late reverberation components of the test signals as captured by
the microphone.
42. The method of claim 34, wherein the acoustic room parameters
are determined in-situ during speech enhancement operation, wherein
a reverberation time of the room is estimated from captured voice
signals, and wherein the acoustic room parameters are derived from
the determined reverberation time.
43. The method of claim 26, wherein the captured audio signals are
transmitted via a wireless link to the audio signal processing
unit.
44. The method of claim 26, wherein the reverberation level is a
late reverberation level corresponding to a level of the components
of the sound generated by the loudspeaker arrangement having
reverberation times above a reverberation time threshold, which
threshold is selected such that late reverberation sound components
are perceivable as a hearing sensation separate from perception of
respective non-delayed sound.
45. The method of claim 44, wherein the reverberation threshold
time is about 50 ms.
46. A system for speech enhancement in a room, comprising a
microphone for capturing audio signals from a speaker's voice, an
audio signal processing unit for processing the captured audio
signals a loudspeaker arrangement to be located in the room for
generating sound according to the processed audio signals, and
means for estimating an ambient noise level in the room from the
captured audio signals, wherein the audio signal processing unit
comprises means for estimating a reverberation level and means for
determining a gain to be applied to the captured audio signals by
the audio signal processing unit according to a comparison between
the estimated ambient noise level and an estimated reverberation
level, wherein the reverberation level is the level of reverberant
components of the sound generated by the loudspeaker
arrangement.
47. The system of claim 46, wherein the system comprises a power
amplifier for amplifying, at constant gain, the processed audio
signals in order to produce amplified processed audio signals to be
supplied to loudspeaker arrangement.
48. The system of claim 47, wherein said means for estimating is
adapted to estimate the reverberation level from a level of the
processed audio signals prior to supplying thereof to the
loudspeaker arrangement as the amplified processed audio
signals.
49. The system of claim 46, wherein the microphone forms part of a
transmission unit comprising a voice activity detector for
analyzing the captured audio signals for outputting a voice
activity status signal indicating whether the speaker is presently
speaking or not, an ambient noise level estimator for estimating
said ambient noise level and for outputting an ambient noise level
signal indicating the estimated ambient noise level, and a
transmitter for transmitting the captured audio signals, the voice
activity status signal and the ambient noise level signal via a
wireless link to a receiver unit comprising a receiver for
receiving the signals transmitted by transmitter and the audio
signal processing unit.
50. The system of claim 49, wherein the transmission unit is
compatible with hearing aids having a wireless audio interface.
Description
BACKGROUND OF THE INVENTION
[0001] 1. Field of the Invention
[0002] The present invention relates to a system for speech
enhancement in a room comprising a microphone for capturing audio
signals from a speaker's voice, an audio signal processing unit for
processing the captured audio signals and a loudspeaker arrangement
located in the room for generating amplified sound according to the
processed audio signals.
[0003] By using such a system, the speaker's voice can be amplified
in order to increase speech intelligibility for persons present in
the room, such as the listeners in an audience or pupils/students
in a classroom. However, increased amplification does not
necessarily result in increased speech intelligibility.
[0004] 2. Description of Related Art
[0005] U.S. Pat. No. 7,333,618 B2 relates to a speech enhancement
system comprising, in addition to the speaker's microphone, a
second microphone placed in the audience for capturing both the
sound generated by the loudspeakers and ambient noise, a variable
amplifier and an ambient noise compensation circuit. The output
signal of the variable amplifier is compared to the ambient noise
level derived from the signals captures by the second microphone,
and the gain applied to the signals from the speaker's microphone
is adjusted according to the level of the ambient noise.
[0006] European Patent Application EP 1 691 574 A2 relates to an FM
(frequency modulation) transmission system for a hearing aid,
wherein the gain applied to the audio signals captured by the
microphone of the FM transmission unit is adjusted in the FM
receiver according to the ambient noise level and the voice
activity as detected by analyzing the audio signals captured by the
microphone. The gain is automatically increased when as it is
detected that the speaker is speaking; the gain is also adjusted as
a function of ambient noise level.
SUMMARY OF THE INVENTION
[0007] It is an object of the invention to provide for a speech
enhancement system, whereby speech intelligibility is increased in
an efficient manner. It is also an object to provide for a
corresponding method of speech enhancement.
[0008] According to the invention, these objects are achieved by a
speech enhancement method and speech enhancement system as
described herein.
[0009] The invention is beneficial in that, by determining the gain
to be applied to the audio signals captured by the microphone
according to a comparison between an estimated ambient noise level
and an estimated reverberation level of the sound generated by the
loudspeaker arrangement, the signal to noise ratio (SNR) can be
optimized at an any time, without applying an unnecessary high
gain, thereby increasing speech intelligibility in an efficient
manner.
[0010] Preferably, the reverberation level is a late reverberation
level corresponding to the level of the components of the sound
generated by the loudspeaker arrangement having reverberation times
above a reverberation time threshold, which threshold is selected
such that the late reverberation sound components are perceivable
as a hearing sensation separate from perception of the respective
non-delayed sound. For example, the reverberation threshold time
may be about 50 ms
[0011] These and further objects, features and advantages of the
present invention will become apparent from the following
description when taken in connection with the accompanying drawings
which, for purposes of illustration only, show several embodiments
in accordance with the present invention.
BRIEF DESCRIPTION OF THE DRAWINGS
[0012] FIG. 1 is a schematic block diagram of a speech enhancement
system according to the invention;
[0013] FIG. 2 is a diagram showing the levels of the useful signal,
the late reverberation signal and the ambient noise signal in a
condition when the gain of the speech enhancement system is too
low;
[0014] FIG. 3 is a diagram like FIG. 2, wherein a condition is
shown when the gain of the speech enhancement system is
optimal;
[0015] FIG. 4 is a diagram like FIGS. 2 and 3 showing a condition
when the speaker is not speaking;
[0016] FIG. 5 is a diagram like FIG. 4 showing a condition when the
speaker starts to speak;
[0017] FIG. 6 is a diagram like FIG. 4 showing a condition when the
ambient voice level changes with time;
[0018] FIG. 7 is a diagram like FIG. 4 showing a condition when the
beginning of feedback has been detected;
[0019] FIG. 8 is a block diagram of an example of a speech
enhancement system according to the invention;
[0020] FIG. 9 is a block diagram of an alternative example of a
speech enhancement system according to the invention;
[0021] FIG. 10 is a block diagram of a further alternative example
of a speech enhancement system according to the invention;
[0022] FIG. 11 is a block diagram of a still further alternative
example of a speech enhancement system according to the invention;
and
[0023] FIG. 12 is a block diagram like FIG. 8, wherein a modified
version is shown.
DETAILED DESCRIPTION OF THE INVENTION
[0024] FIG. 1 is a schematic representation of a system for
enhancement of speech in a room 10. The system comprises a
microphone 12 (which in practice may be a directional microphone
comprising at least two spaced apart acoustic sensors) for
capturing audio signals from the voice of a speaker 14, which
signals are supplied to a unit 16 which may provide for
pre-amplification of the audio signals and which, in case of a
wireless microphone, includes a transmitter for establishing a
wireless audio signal link, such as an analog FM link or,
preferably, a digital link. The audio signals are supplied, either
by cable or in case of a wireless microphone, via an audio signal
receiver 18, to an audio signal processing unit 20 for processing
the audio signals, in particular to apply spectral filtering and
gain control to the audio signals. The processed audio signals are
supplied to a power amplifier 22 operating at constant gain in
order to supply amplified audio signals to a loudspeaker
arrangement 24 in order to generate amplified sound according to
the processed audio signals, which sound is perceived by listeners
26.
[0025] The purpose of a speech enhancement system in a room is to
increase the intelligibility of the speaker's voice. In general,
speech intelligibility is affected by the noise level in the room
(ambient noise level) and the reverberation of the useful sound,
i.e., the speaker's voice, in the room. At least part of the
reverberation acts to deteriorate speech intelligibility. The total
reverberation signal may be split into an early reverberation
signal (corresponding to reverberation times of e.g. not more than
50 ms) and a late reverberation signal (corresponding reverberation
times of more than 50 ms). The early reverberation signal is
integrated with the direct sound by the human hearing, i.e., it is
not perceivable as a separate signal, and therefore does not
deteriorate speech intelligibility. The late reverberation signal
is not integrated with the direct sound by the human hearing, it is
perceivable as a separate signal, and therefore has to be
considered as part of the noise.
[0026] Hence, the acoustic field in a room may be separated into
three parts: (1) the useful signal, i.e., the direct field of the
speaker's voice and the respective early reverberation signal; (2)
the late reverberation signal, e.g. the reverberation signal of the
speaker's voice corresponding reverberation times of more than 50
ms; (3) the ambient noise, i.e., the noise from all other sources.
By "speaker's voice," here, the speaker's voice as reproduced by
the loudspeaker arrangement 24 is meant.
[0027] When the gain applied in the audio signal processing unit 20
is increased, both the level of the "useful signal" and the level
of the "late reverberation signal" will increase, whereas the level
of the "ambient noise" is independent of the speaker's voice level
and hence will not increase when the gain is increased. However, of
course, the ambient noise level may vary in time when, for example,
some of the listeners 26 start talking, etc.
[0028] FIG. 2 is a schematic representation of these three sound
field components, wherein the level of the late reverberation
signal is lower than the ambient noise level. In this case the
signal to noise ratio (SNR), which is a measure of the speech
intelligibility, is determined by the difference between the level
of the useful signal and the ambient noise level.
[0029] As shown in FIG. 3, the SNR can be increased by increasing
the gain applied to the audio signals captured by the microphone
12, because thereby the level of the useful signal is increased,
while the ambient noise level remains constant.
[0030] However, since the level of the late reverberation signal
increases in parallel with the level of the useful signal, a
further increase in gain will not result in a corresponding
increase in SNR once the ambient noise is masked by the late
reverberation signal. It can be assumed that such masking of the
ambient noise occurs when the level of the late reverberation
signals is at least about 3 dB higher than the level of the ambient
noise. This situation is shown in FIG. 3, according to which the
SNR is optimized when the gain is set to a value at which the level
of the late reverberation signal is about 3 dB higher than the
ambient noise level. As already mentioned above, further increase
of the gain then will not result in an increase in SNR and hence
should be avoided.
[0031] In order to optimize the gain (and hence the SNR), it is
beneficial to estimate both the actual level of a reverberation
signal, which is preferably the late reverberation signal discussed
above, and the actual level of the ambient noise.
[0032] The threshold of the reverberation time from which on the
sound components form part of the (late) reverberation level
preferably is selected such that the late reverberation sound
components are perceivable as a hearing sensation separate from the
perception of the respective non-delayed sound. The threshold in
practice corresponds to that reverberation time at which a sound
component starts to create a hearing sensation perceived separately
from that of the respective non-delayed signal. Typically, the
threshold may be set at around 50 ms.
[0033] Whereas the ambient noise level is estimated from the audio
signals captured by the microphone 12, the (late) reverberation
level may be estimated either from the level of the processed audio
signals, namely the level of the audio signals at the input of the
power amplifier 22, (closed loop configuration) or from the level
of the audio signals supplied to audio signal processing unit 20,
i.e., from the level of the audio signals prior to being processed
(open loop configuration).
[0034] Typically, gain changes slowly, with time constants on the
order of about 5 s.
[0035] In FIG. 8, a first example of a speech enhancement system
according to the invention is shown, wherein the system is designed
as a wireless system, i.e., comprising a wireless audio link,
preferably a digital link, for transmitting the audio signals from
the microphone 12 to the loudspeakers 24. The system comprises a
transmission unit 16 including the microphone 12, a voice activity
detector (VAD) 32, an ambient noise level estimator 34 and an RF
(Radio Frequency) transmitter 36, which may be digital.
[0036] The voice activity detector 32 analyzes the audio signals
captured by the microphone 12 and determines whether the speaker 14
is presently speaking or not and outputs a corresponding VAD status
signal. The ambient noise level estimator 34 is active only when
the VAD signal supplied from the voice activity detector 32
indicates that the speaker 14 presently is not speaking. The
ambient noise level estimator 34, when active, derives from the
audio signals captured by the microphone 12, an ambient noise
compensation (SNC) signal, which is indicative of the present
ambient noise level.
[0037] The audio signals captured by the microphone 12, the VAD
signal and the SNC signal are supplied to the transmitter 36 for
being transmitted via a radio frequency (RF) link, such as an FM
link, to an RF receiver 18, which supplies the received signals to
the audio signal processing unit 20 which comprises a feedback
canceller 38, a SNR optimizer 40, a late reverberation level
estimation unit 42 and an automatic gain control unit 44. The audio
signals received by the receiver 18 are supplied via the feedback
canceller 38 to the automatic gain control unit 44, in order to be
transformed into processed audio signals which are supplied as
input to the power amplifier 22 which drives the loudspeaker
arrangement 24. The late reverberation level estimation unit 42
uses the level of the processed audio signal supplied by the
automatic gain control unit 44 to the power amplifier 22 for
estimating the late reverberation level by taking into account
acoustic room parameters.
[0038] In the embodiment of FIG. 8, the acoustic room parameters
are fixed, i.e., factory-programmed, and are that of a typical room
in which the loudspeaker arrangement 24 is to be used. Preferably,
the late reverberation level is estimated by applying a correction
factor derived from the acoustic room parameters to a level
measurement of the audio signals at the input of the power
amplifier 22.
[0039] The feedback canceller 38 analyses the audio signals
received by the receiver 18 in order to determine whether there is
a critical feedback level caused by feedback of sound from the
loudspeaker arrangement 24 to the microphone 12 (Larsen effect), As
a result the feedback canceller 38 outputs a status signal
indicating the presence or absence of critical feedback, which
status signal is supplied to the SNR optimizer 40, together with a
signal indicative of the late reverberation level estimated by the
unit 42 and the SNC and VAD signals received by the receiver 18.
Based on the information provided by these input signals, the SNR
optimizer 40 outputs a control signal acting on the automatic gain
control unit 44 for controlling the gain, in order to optimize the
SNR, as will be illustrated by reference to FIGS. 4 to 7.
[0040] During times when the VAD signal indicates that the speaker
14 is not speaking, the ambient noise estimator 34 determines the
ambient noise level (SNC-signal) from the audio signals presently
captured by the microphone 12. This situation is shown in FIG. 4;
at the position of the listeners 26 the ambient noise is
dominant.
[0041] During times when the VAD signal indicates that the speaker
14 is speaking, the gain is increased to the ambient noise level
expected to be masked by the late reverberation level. For example,
the gain may be increased until the late reverberation level is
about 3 dB above the ambient noise level, see FIG. 5.
[0042] When the ambient noise level estimator 34 determines that
the ambient noise level has changed, the gain will be adjusted by
the SNR optimizer 40, with a certain time constant, to the
presently estimated ambient noise level. In other words, when the
ambient noise level is found to decrease, the gain is decreased
accordingly, and when the ambient noise level is found to increase,
the gain is increased accordingly, see FIG. 6. Thereby, the SNR can
be optimized at any time.
[0043] However, for high ambient noise levels it might be necessary
to increase the gain to a value at which the system starts to have
feedback problems. Once such condition is determined by the
feedback canceller 38, a further increase of the gain will be
stopped by the SNR optimizer. Under such conditions, the ambient
noise level may become higher than the late reverberation level, so
that the SNR then will be lower than at lower ambient noise levels,
see FIG. 7.
[0044] While FIG. 8 shows an embodiment having a closed loop
configuration (the late reverberation level is determined from the
processed audio signals at the output of the automatic gain control
unit 44), FIG. 12 shows the embodiment of FIG. 8 as modified to an
open loop configuration, wherein the reverberation level is
determined from the (non-processed) audio signals at the input to
the automatic gain control unit 44.
[0045] In FIG. 9, the block diagram of another modified system is
shown, wherein, for estimating the late reverberation level,
acoustic parameters of the actual room in which the system is used
are determined from a measurement carried out in a calibration mode
prior to using the system for speech enhancement. According to the
embodiment of FIG. 9, the acoustic room parameters are determined
by measurement of the level of the reverberant field in the room.
To this end, the user places the microphone 12 at a position in the
room 10, which position is dominated by the reverberant sound from
the loudspeaker arrangement 24, and launches an automatic
calibration procedure. According to the embodiment of FIG. 9 the
late reverberation level estimation unit 42 of the embodiment of
FIG. 8 is replaced by a unit 142 which serves to both determine the
acoustic parameters of the room and to estimate the late
reverberation level.
[0046] In the calibration mode, the unit 142 generates a test
signal which is supplied via the power amplifier 22 to the
loudspeaker arrangement 24 for reproducing a corresponding test
sound which is captured by the microphone 12 as test audio signals
from which the SNC signal, which corresponds to the level of the
test sound, is derived by the ambient noise level estimator 34,
with the SNC signal being supplied to the unit 142. The unit 142
analyzes the SNC signal corresponding to the test signal level, and
a ratio of the level of the signal at the input of the power
amplifier 22 and the test audio signal level determined by the unit
142 is calculated and stored in a memory 146 connected to the unit
142.
[0047] In other words, in the calibration mode, a test signal
having a known level is generated via the loudspeaker arrangement
24, the test signal is captured by the microphone 12, and the
correction factor to be applied to the level of the processed audio
signals at the input of the power amplifier 22 in order to estimate
the late reverberation level is determined from the level of the
test audio signals captured by the microphone 12. In the speech
enhancement mode of the system, the correction factor us retrieved
from the memory 146.
[0048] The system of FIG. 9 is an open loop system, i.e., like in
the system of FIG. 12, the reverberation level is determined from
the (unprocessed) audio signals at the input to the automatic gain
control unit 44.
[0049] In FIG. 10, an embodiment is shown wherein, in the
calibration mode, the acoustic room parameters are determined by
measurement of the impulse response of the room 10 rather than by
measurement of the level of the reverberant field in the room 10 as
realized in the embodiment of FIG. 9. In this case, in the
calibration mode the microphone 12 may be placed at any position in
the room, and the unit 142 generates a maximum length sequence
(MLS) test signal at a known level, which is supplied via the power
amplifier 22 to the loudspeaker arrangement 24 for reproducing a
corresponding test sound which is captured by the microphone 12.
The captured test audio signals are supplied via the wireless link
to the unit 142. In the unit 142, a convolution of the captured
test audio signals is performed in order to obtain the impulse
response of the system in the room 10, wherein only the level of
the late reverberation sound components, e.g., test sound
components corresponding to reverberation times of more than 50 ms,
are taken into account.
[0050] In other words, the correction factor to be applied to the
level of the processed audio signals at the input of the power
amplifier 22 is determined from the level of the late reverberation
components of the test audio signals as captured by the microphone
12. To this end, a ratio of the audio signal level at the input of
the power amplifier 22 (i.e., the level of the processed test audio
signals) and the late reverberation level of the test audio signals
as measured by the unit 142 is calculated and stored in the memory
146. In the speech enhancement mode, the value stored in the memory
146 then is used to estimate the late reverberation level from the
audio signal level at the input of the power amplifier 22.
[0051] Although the system of FIG. 10 is shown as a closed loop
system, alternatively, it could be designed as an open loop
system.
[0052] In FIG. 11, an embodiment is shown wherein an in-situ
determination of the acoustic parameters of the actual room 10, in
which the system is used, is enabled during speech enhancement
operation, without a calibration mode being necessary. In this
case, the transmission unit 16 includes a reverberation time
estimation unit 30, which is able to determine a reverberation time
of the room, such as RT60, from the audio signals captured by the
microphone 12 during speech enhancement operation, i.e., when the
speaker 14 is speaking (RT60 is the time needed for the reverberant
field in the room to decrease by 60 dB after an impulse noise;
usually, RT60 is determined as a function of frequency). The RT60
value determined by the reverberation time estimation unit 30 is
supplied to the transmitter 36 for being transmitted via the
receiver 18 to the SNR optimizer 40. The SNR optimizer 40 creates a
set of acoustic room parameters according to the RT60 measurement
and estimates the late reverberation level by using a corresponding
correcting factor applied to the level of the processed audio
signals at the input of the power amplifier 22.
[0053] Although the system of FIG. 10 is shown as a closed loop
system, alternatively, it could be designed as an open loop
system.
[0054] In all embodiments, the transmission unit 16 may be
compatible with hearing aids having a wireless audio interface,
such as hearing aids having an FM receiver unit connected via an
audio shoe to the hearing aid or hearing aids having an integrated
FM receiver.
[0055] While various embodiments in accordance with the present
invention have been shown and described, it is understood that the
invention is not limited thereto, and is susceptible to numerous
changes and modifications as known to those skilled in the art.
Therefore, this invention is not limited to the details shown and
described herein, and includes all such changes and modifications
as encompassed by the scope of the appended claims.
* * * * *